similar to: Mystery phone!

So I think I am able to reproduce the dropped call scenario. Here is what I do to get a dropped call: 1. Call 1-800-tmobile 2. Go true their IVR and get connected to the customer service IVR 3. Enter my number and SSN 4. press 0 5. Then the audio please hold starts. After about 2-4 seconds the call gets dropped. (fast busy tone) The time on my phone will stop running (call time) and I will get

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as the

How can this scenario be implemented please? THIS IS NOT A SEND TEXT application. A call arrives on the IVR. After hearing several vectors to guide the person through information I want to confirm a transaction via email to his cell phone. Specifically, I want to use his phone number and then append the SMTP suffix from his service provider. Press 1 if you use Verizon, 2 if you use ATT, 3 if

I'm setting rsync up for the first time and would prefer to have the exclude file defined in the conf file, but the exclusions aren't honoured when I define the parameter in rsyncd.conf - although they are when I specify the file in an argument. The server is the remote system and both rsyncd.conf and the exclude file are the same on both local and remote systems. I'm attaching the

Please see the attached file and let me know if you need any more information. /Sam Sam Sexton <mailto:sam.sexton@reuters.com> Reuters Coventry Automated Dealing Technologies Phone: +44 24 7625 6562 Fax: +44 24 7655 5203 --------------------------------------------------------------- - Visit our Internet site at http://www.reuters.com Get closer to the

Created this function on one of my machines today, thought others might find it useful: freesip() { comm -2 <(seq $2 $3) <(cat $1 | grep ^\\[ | sort | uniq | tr -d \[ | tr -d \]) | grep ^[[:digit:]] } On RedHat/CentOS based systems you can create the following file to have the function available on login: /etc/profile.d/freesip.sh # Free SIP extensions freesip() { comm -2 <(seq $2

Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) -- Kyle Sexton

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

On 2/5/2015 12:56 AM, Steffen Kaiser wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Wed, 4 Feb 2015, George Sexton wrote: > >> I'm seeing two bogus messages appearing my Maildir/cur directory. >> They're dovecot.index.log and dovecot-uidlist. >> >> -rw------- 1 gsexton users 51 Feb 4 09:04 >> Maildir/cur/dovecot-uidlist:2,S

I'm seeing two bogus messages appearing my Maildir/cur directory. They're dovecot.index.log and dovecot-uidlist. -rw------- 1 gsexton users 51 Feb 4 09:04 Maildir/cur/dovecot-uidlist:2,S -rw------- 1 gsexton users 244 Feb 4 09:04 Maildir/cur/dovecot.index.log These files are only appearing in the Maildir/cur directory, and not in any other directory that mail is delivered to (e.g.

Having a problem w/ not getting CID name from a PRI. CID Name appears in the PRI debug, but even after a Wait(4) it still appears after the phone is ringing. Here is the relevant info from my PRI debug output. Line 4 is a NoOp showing me trying to echo Name and Number. Line 6 dials the extension, and you can see callerid name get presented on line 29. Again, there is a Wait(4) before the

All, I am experiencing an issue where if an agent is logged into the queue, but has their client closed. It appears that when the queue calls the agent, it goes through the macro I have setup for that user and will dump them to voicemail if unavailable. This pulls the call out of the queue, which is not what I would like to happen. I am wondering if this is the expected behavior and I should

I have two servers setup to do DUNDi lookups against each other. The scenario is that on server A, I have a wildcard match for extensions 64XX that rings to a local extension on the server. On server B I have a 6442 real extension that I would like to have ring if called. It seems that DUNDi is matching on the 64XX and not searching out to see if there is a *more* exact match than the pattern

Dear Gavin, What is the interpretation of a simulated p-value of 1 in a mantel test and how is the p-value derived? When we run two highly negatively correlated matrices we get this result: r: -1, simulated p value: 1 I would expect the high p-value of 1 to mean not significant. Could you clarify? Thank you, Jason Jason P. Sexton Graduate Group in Ecology University of California,

Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * AT&T/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __________________________________ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com

*Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller* Equip the Asterisk server